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A FUNDAMENTAL INTRODUCTION

TO THE COMPACT DISC PLAYER

Grant M. Erickson
Department of Electrical Engineering
University of Minnesota

November 29, 1994


EE 3011
Professor: Dr. Kevin M. Buckley
TABLE OF CONTENTS

THE NEED FOR DIGITAL AUDIO.............................................................. 1


Strengths of the Digital Domain ......................................................... 1
Developments Facilitating the Compact Disc Player ............................. 2
PRINCIPLES OF DIGITAL AUDIO ............................................................ 3
Sampling......................................................................................... 3
Aliasing .......................................................................................... 4
Quantization.................................................................................... 5
Dither ............................................................................................. 6
Jitter............................................................................................... 7
IMPLEMENTATION................................................................................. 8
System Overview .............................................................................. 8
Digital-to-analog Converters .............................................................. 10
Multi-bit Converters................................................................. 10
Low-bit Converters................................................................... 11
Digital Filtering, Oversampling, and Noise Shaping ............................ 14
REFERENCES........................................................................................... 19

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LIST OF FIGURES

Table 1. Red Book specifications for the compact disc system....................... 9


Figure 1. Block diagram of a compact disc player. ....................................... 9
Figure 2. Block diagram of a PWM/MASH digital-to-analog converter. .......... 12
Figure 3. Block diagram of a PDM digital-to-analog converter....................... 13
Figure 4. Use of a transversal filter to achieve oversampling ........................ 15
Figure 5. Effect of zeros interleaving and oversampling on a signal. ............. 15
Figure 6a. Various noise-density distributions as a function of
frequency. . ............................................................................... 16
Figure 6b. The PDM and PWM distributions in the audio band. ..................... 16
Figure 7. Third-order noise shaper used in the PWM converter. ................... 17

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A Fundamental Introduction to the Compact Disc Player

The compact disc player has become one of the most ubiquitous pieces of
consumer electronics equipment in use today. Tens of millions of players have
been sold to date. However, as pervasive as the compact disc players presence is,
the beauty and complexity of its design and operation are underappreciated by
most users. This brief text attempts to inform the reader of the basic
fundamentals of the compact disc player. It is a assumed that the reader has a
basic knowledge of the fundamentals of signal processing, although it is certainly
not a prerequisite for learning a great deal from the reading.

THE NEED FOR DIGITAL AUDIO

Strengths of the Digital Domain


Since Thomas Edison made the first audio recording on a foil covered cylinder in
1877, the field of audio recording has grown and matured. Edisons process and
many others that followed were all based on a common process; the reproduction
of an audio signal from a mechanical or electrical contact with the recording
mediathis is the realm of analog audio. After nearly 100 years, analog audio
has reached a mature state and nearly all of its shortcomings have been
addressed to the point that further improvements become financially prohibitive
for the average consumer.

The very nature of the analog signal leads to its own shortcomings. In the
analog domain, any waveform is allowable; therefore the playback mechanism
has no means to differentiate noise and distortion from the original signal.
Further, in an analog system every copy made introduces more noise than its
parent. This fact is due to both the playback and recording mechanism which
must physically contact the media, further damaging it after every pass. Every
analog system also carries the side effect that the total system noise is the
summation of all distortion and noise from each component in the signal path.
Finally, analog equipment is of limited performance, exhibiting: an uneven
frequency response (which requires extensive equalization), a limited 60 dB
dynamic range, and a 30 dB channel separationwhich affects stereo imaging
and staging.

1
2

The need for a new audio format is apparent, and digital audio fills the
performance shortcomings of analog audio. The beauty of the digital audio signal
is that noise and distortion can be separated from the audio signal. A digital audio
signals quality is not a function of the reading mechanism nor the media in a
properly engineered system. Performance parameters such as frequency
response, linearity, and noise are only functions of the digital-to-analog converter
(DAC). Performance parameters indicative of a digital audio system include full
audio band frequency response of 5 ~ 22,000 Hz, 90+ dB dynamic range, and a flat
response across the entire audio band.

The final strength of digital audio is the circuitry upon which it is built. First,
due to a large degree of circuit integration digital circuits do not degrade with
time as analog circuits do. Further, for all practical purposes, a digital signal will
suffer no degradation until distortion and noise has become so great that the
signal is out of its voltage threshold. However, this threshold has been made
intentionally large expressly for this reason. The high level of circuit integration
also means that for the same given task, the digital circuitry will cost far less
than its analog counterpart.

The only real theoretical limitation to the accuracy of a digital signal is the
quantity of numbers in the signal representation and the accuracy of those
numbers. These are both known and controllable design parameters.

Developments Facilitating the Compact Disc Player


As staggering as the release of the compact disc player was in 1982, the
technology and theories which allowed it to be born were long in development. In
1841, the great mathematician Augustin-Louis Cauchy first proposes the
sampling theorem. Nearly 80 years later J.R. Carson publishes a mathematical
analysis of time sampling in communications. In a 1928 lecture at the American
Institute of Electrical Engineers Harry Nyquist provides proof of the sampling
theorem in Certain Topics in Telegraph Transmission Theory. In 1937, A.
Reeves proposes pulse code wave modulation (PCM). In 1948, John Bordeen,
William Shockley, and Walter Brattain invent the bipolar junction transistor at
Bell Labscompact digital circuitry is a reality. Two years later, in 1950 Richard
W. Hamming publishes significant work on error correction and detection codes.
In 1958 C.H. Townes and A.L. Shawlow invent the laser. In 1960 R.C. Bose
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publishes binary group error correction codes. That same year I.S. Reed and G.
Solomon publish error correction codes to be used in the CD player 22 years later.
Also early computer music experiments take place at Bell Labs. Fifteen years
before consumers see the first player, NHK Technical Research Institute publicly
demonstrates a PCM digital audio recorder with a 30 kHz sampling rate and 12-
bit resolution. Two years later, Sony Corporation demonstrates a PCM digital
audio recorder with a 47.25 kHz sampling rate and 13-bit resolution. A
hemisphere away, Dutch physicist Klaas Compaan uses a glass disc to store black
and white holographic images using frequency modulation at Philips
Laboratories. Four years later, in 1973 Philips engineers begin to contemplate an
audio application for their video disc system. A prototype disc with a 44 kHz
sampling rate is run through a 14-bit digital-to-analog converter and exhibits a
signal-to-noise (S/N) ratio of 80 dB in monaural. Now a research frontier,
Mitsubishi, Sony, and Hitachi all demonstrate digital audio discs at the Tokyo
Audio Fair in 1977. One year later, Philips joins with its recording subsidiary
Polygram Records to develop a worldwide digital audio standard. In March 1979,
Philips demonstrates a prototype compact disc player in Europe. Sony joins the
Philips/Polygram coalition after Matsushita declines. In June of 1980, the
coalition formally proposes their CD standard. A year later in 1981, Sharp
successfully mass produces the semiconductor laser. This step was crucial to
delivering a consumer product. In Fall of 1982 nearly 150 years of work comes to
fruition and Sony and Philips introduce their respective players to consumer in
Europe. The following spring, the player is introduced in the United States.
Twelve years later, the improvement of digital audio continues at a rapid pace and
the analog format that was so prevalent in 1982 has all but disappeared.

PRINCIPLES OF DIGITAL AUDIO

Sampling
Given an analog audio signal, a process is needed to bring it into the digital
domain. This process is sampling, and it is dictated by the Nyquist sampling
theorem which states how quickly samples must be taken to ensure an accurate
representation of the analog signal.

The sampling theorem is quite simple. It states that the sampling frequency
must be greater than or equal to the highest frequency in the original analog
4

signal. The relationship is given by Equation 1; note that the theorem can also be
expressed in terms of the sampling period.
T
f s 2 f or Ts (1)
2

The sampling theorem is simple enough, but to use it in a digital audio system,
two constraints must be observed. The first is that the original signal must be
bandlimited to half the sampling frequency by being passed through an ideal low-
pass filter; the second is that the output signal must again be passed through an
ideal low-pass filter to reproduce the analog signal. These constraints are crucial
to sampling, and if not observed will lead to an unwanted effect known as
aliasing.

Aliasing
Aliasing is a systems erroneous response that manifests itself when the
constraints of the sampling theorem are not observed. Aliasing will surface in the
audio signal as audible distortion. For the limiting case of a frequency at exactly
half the sampling frequency, there will be only two samples generatedthis is the
f
minimum required to represent any waveform. For signals greater than s 2 , the
process of sampling can be thought of as modulating the input signal. The
modulation creates image frequencies centered around integer multiples of fs .
These newly generated frequencies are then imaged or aliased back into the
audible band. The frequency to which these will be aliased to can be computed by
Eq. 2, where fa is the alias frequency, f is the actual frequency, fs is the sampling
frequency, and k is an odd integer that satisfies the inequality.

fa = f
( k + 1) f s kf s f ( k + 2 ) f s (2)
2 2 2

We can then easily compute for a sampling rate of 44.1 kHz, a signal of 23 kHz will
be aliased to 21.1 kHz. More precisely, the frequency will be folded back across half
the sampling frequency by the amount it exceed half the sampling frequencyin
this case by 950 Hz.

Hence the use of a brickwall filterone with a sharp cutoff characteristicon


the input signal is necessary. The need for placing a filter after the DAC in the
player may not be intuitively obvious. Imagine the limiting case of a sine wave at
half the sampling frequency. There will be two samples generated for this wave,
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however the DAC will represent this as a square wave of the same frequency.
From the Fourier series expansion, we know that a square wave consists of
infinite harmonics. The DAC has now created frequencies that did not previously
exists. Because the input signal was bandlimited, we know that it is reasonable to
pass the output signal through another low-pass filter with the same
characteristic as that used in the sampling process. This low-pass filter strips the
higher-order harmonics from the square wave and we are left with the sine wave
we started with. Due to its actions, this low-pass filter is often referred to as an
anti-aliasing filter in the frequency domain and as a reconstruction filter in the
time domain. A linear phase low-pass filter is characterized by having a
symmetrical impulse response. In particular, the impulse response of a low-pass
filter is the sin(x)/x function. When the reconstruction filter is excited by an
amplitude varying impulse train from the DAC, the output is a linear
combination of the individual amplitude modulated impulse responses.

Quantization
Once sampling has taken place, we are far from done converting the analog
signal to a digital one. In order to represent each sample as a binary series of bits,
the infinitely varying voltage amplitude of the analog signal must be assigned a
discrete value. This process of assignment is known as quantization. It is
important to note that quantization and sampling are complementary processes.
If we sample the time axis, then we must quantize the amplitude axis and vice
versa. It is unfortunately common practice to refer to sampling and quantization
as just quantization; this is, however, incorrect. The combined process is referred
to as digitization.

In a 16-bit audio format, we can represent a sinusoidally varying voltage audio


signal by 216 or 65,536 discrete levels. It is apparent then that quantization is a
limiting performance factor in the overall digital audio system, by the number of
bits allowed to the quantizing system. The system designer is faced with
determining how many bits create a sufficient model of the original signal.

Because of this limiting design factor, quantizing is ideally imperfect in its


signal representation, whereas sampling is theoretically perfect. There is then an
error inherent in the quantization process regardless of the ideality of the rest of
the system.
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To visualize what this error is, imagine a digital thermometer on your oven.
When the temperature reads 425 F, that value may or may not be accurate. The
temperature in the oven may indeed be 425, but it might also be as much as 425.4
or as little as 424.5. A similar occurrence occurs with the quantizer in digital
audio equipment. While quantizing, it determines the level in which the voltage
Q
for a given sample belongs. This quantized level may differ by as much as ,
2
where Q is the width of the quantized level.

It is the difference between the actual voltage to be represented and the


quantized voltage level that induces quantization error. The magnitude of the
error may never exceed the voltage represented by half of the least-significant bit
(LSB) in the data word. A measurement of the error in a digitization system can
be made, and it is expressed as the signal-to-error (S/E) ratio. This ratio is given
by Eq. 3, where n is the number of bits in the data word.
S E ( dB ) = 6.02n + 1.76 (3)

Hence, the theoretical S/E ratio for a 16-bit system is 98 dB. Keep in mind that this
value is strictly theoretical and will be lowered and raised by many other
performance parameters. For the most part, quantization error manifests itself
as noise at high signal levels. However, quantization error becomes quite
significant when a low-level signal approaches the level of the LSB, then the
quantizing error actually becomes the signal, and therefore is an audible
component of the output. In many types of music, these types of signals are
common and distortion caused by quantization error is both unacceptable and
irremovable. Fortunately, in practical systems this adverse effect can be effectively
eliminated through the use of dither.

Dither
Dither is the process of adding low-level analog noise to a signal, to randomize
or confuse the quantizers small-signal behavior. Dither specifically aims to
address two problems in quantization. The first of which is that a reverberating,
decaying signal can fall below the lower limit of the system resolution. That is to
say that an attempt to encode a signal below the LSB results in nothing getting
encoded. Clearly, information is lost. The second, as discussed in the previous
section, is that system distortion increases as a percent of a decreasing input
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signal. It is important to note that not only does dither remove some quantization
error from the signal, it effectively removes it.

The concept might seem initially counterintuitive, but it is really quite simple.
Dither relies on some special behavior of the human ear. The ear can detect a
signal masked by particularly broadband noise. In some cases, the ear can easily
detect a midrange signal buried as much as 10 to 12 dB below the level of
broadband noise1 . Those who still find the effects of dither questionable, might
want to try the following interesting test2.

Let the text on this page represent the amplitude of the signal to be quantized.
Also, let the space between your slightly spread fingers represent valid
quantization intervals. Now place your hand across the text. Amplitude
information has been irrecoverably lost due to quantization. Now provide dither to
the signal by quickly moving your hand up and down along the plane of the page.
The amplitude information that was lost has been retrieved at the expense of
adding a slight amount of noise to the systemyour blurred fingers.

So even though some noise has been added, we have eliminated the distortion
due to quantization error with the result being a cleaner, more accurate signal.

Jitter
Although rarely observed in a well designed player, jitter is a worthy topic of
discussion because of both its misconceptions and the large amount of press it has
received. Jitter is basically defined as time instability. It occurs in both analog-to-
digital and digital-to-analog conversion. The latter instance is the only concern
here. Jitter occurs in the compact disc player when samples are being read off the
disc. These reads are controlled by the pulses of a crystal oscillator. If the system
clock pulse inaccurately (an unlikely event), if there is a glitch in the digital
hardware, or if there is noise on a signal control line, the actual reading time will
vary from sample to sample thus inducing noise and distortion in the extreme
case.

A great deal of money has been made by shrewd marketeers preying on the fears
of the consumer worried about jitter. Such products marketed include disc
stabilizer rings to reduce rotational variations, highly damped rubber feet for the
players, and other snake oil remedies. However, the careful engineer has beaten
8

the marketeer to the punch by having the samples read off the disc into a RAM
buffer. As the buffer becomes full, a local crystal oscillator can then clock-out
the samples in a reliable manner, independent of the transport and reading
mechanisms. This process is referred to as timebase correction and as stated
before, any quality piece of equipment will implement it.

IMPLEMENTATION

System Overview
The compact disc player as a sound reproduction device fulfills the loop begun in
the recording studio, returning the audio signal back to its original analog form.
If all the theoretical guidelines have been followed in the equipment and processes
between the musician and your audio system, the sound you hear is exactly the
sound that was heard in the recording studio.

The specifications for the compact disc and compact disc players were jointly
developed by Sony, Philips, and Polygram as mentioned previously. This
specification is contained in their standards document referred to as the Red
Book. A summary of this standard is seen in Table 1.

DISC

Playing time: 74 minutes, 33 seconds maximum

Rotation: Counter-clockwise when viewed from readout surface

Rotational speed: 1.21.4 m/sec. (constant linear velocity)

Track pitch: 1.6 m

Diameter: 120 mm

Thickness: 1.2 mm

Center hole diameter: 15 mm

Recording area: 46 mm 117 mm

Signal area: 50 mm 116 mm

Material: Any acceptable medium with a refraction index of 1.55

Minimum pit length: 0.833 m (1.2 m/sec) to 0.972 m (1.4 m/sec)

Maximum pit length: 3.05 m (1.2 m/sec) to 3.56 m (1.4 m/sec)

Pit depth: ~0.11 m

Pit width: ~0.5 m


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OPTICAL SYSTEM

Standard wavelength: = 780 nm (7,800 )

Focal depth: 2 m

SIGNAL FORMAT

Number of channels: 2 channels (4 channel recording possible)

Quantization: 16-bit linear

Quantizing timing: Concurrent for all channels

Sampling frequency: 44.1 kHz

Channel bit rate: 4.3218 Mb/sec

Data bit rate: 2.0338 Mb/sec

Data-to-channel bit ratio: 8:17

Error correction code: Cross Interleave Reed-Solomon Code (with 25% redundancy)

Modulation system: Eight-to-fourteen Modulation (EFM)

Table 1. Red Book specifications for the compact disc system3.

The compact disc player contains two main subsystems: the audio data
processing system and the servo/control system. The servo, control, and display
system orchestrate the mechanical operation of the player and include such items
as the spindle motor, auto-tracking, lens focus, and the user interface. The audio
data processing section covers all other player processes. A block diagram of the
compact disc player is shown in Figure 1.

LASER PICKUP,
FOCUS, TRACKING, CONTROL AND
SERVO CONTROL DISPLAY
AND DISC TRANSPORT DISPLAY SUBCODE
CONTROL FUNCTIONS FUNCTIONS
DECODING

MEMORY

DIGITAL-TO- LOW-PASS LEFT-


HF SIGNAL INPUT ERROR ANALOG OUTPUT CHANNEL
FROM DISC EFM SAMPLE CIRC ERROR CONCEALMENT CONVERTER FILTER
BIT DETECTION OUTPUT
DEMODULATION RAM BUFFER CORRECTION AND DIGITAL
DEMULTIPLEXING FILTERS
DIGITAL-TO- LOW-PASS RIGHT-
ANALOG OUTPUT CHANNEL
CONVERTER FILTER OUTPUT

SYNCHRONIZATION
CLOCK
DETECTION AND
REGENERATION
TIMING

CRYSTAL
OSCILLATOR

Figure 1. Block diagram of a compact disc player.


10

Since the introduction of the compact disc player in 1982, the market has seen
three generations of players. First generation players were characterized by
multi-bit DACs used with brickwall reconstruction filters. Second generation
players used the same multi-bit DACs but took advantage of digital oversampling
filters placed upstream of the DAC along with a gentle analog reconstruction
filter. Finally, current players make use of low-bit DACs along with
oversampling filters and the gentle analog output filter. In the following sections,
each of these DAC types and filtering methods will be investigated.

Digital-to-analog Converters
The very first demonstration players made by Sony, Philips, and others used 14-
bit converters, which at the time were a vast improvement over analog equipment,
but nonetheless were poor quality by todays standards. By the time the first
consumer players were released in 1982, 16-bit converters were the standard. By
1989, many manufacturers touted the use of 18 and 20-bit converters.

MULTI-BIT CONVERTERSAt the digital hardware level, multi-bit converters may


be designed in several ways. The most common of these include the ladder
network converter, integrating converter, and dynamic element matching
converter. The discussion of these implementations is beyond the scope of this
text, so the ambitious reader is referred to the reference material.

The number of bits in a DAC is a poor method of determining its performance


and accuracy. A better measure of performance is the accuracy of the actual bits
themselves. Under ideal circumstances, a 16-bit converter would exactly convert
all 16-bits of the sample data word in a linear fashion. However, this is seldom
possible. In practice a 16-bit DAC is less than sufficient for accurate conversion.

The error in a 16-bit (or any multi-bit) converter relies on the accuracy of the
most significant bit (MSB) of the data word. Inaccuracy in this bit can result in an
error of half the signals amplitudea significant error by any measure. This in
mind, manufacturers reasoned that converters with high bit rates could
overcome this shortcoming along with others through sheer numbers. In
addition to ensuring the accuracy of the MSB by having more than 16-bits, they
can also improve quantization performance by adding 2x-16 more quantization
11

levels than a 16-bit converter. Now, any nonlinearity in the conversion process
would be a far smaller fraction of the overall signal and the more quantization
levels result in a greater S/E ratio by virtue of Eq. 1. The extra bits used by these
converters may be either thrown away, be left unused, or be put to other intelligent
uses that will be discussed later. Unfortunately, it is a misconception that the use
of an 18- or 20-bit DAC gives true 18 or 20-bit audio performance.

Despite the fantastic performance benefits of these n t h generation multi-bit


converters, they are still plagued by many errors. Linearity was already
mentioned, but they are also plagued by gain error, slew-rate distortion, and zero-
crossing distortion. All of these error and distortion types introduce severe
harmonic distortion and group delay; thereby perturbing signal stability,
imaging, and staging.

Two methods of output reconstruction have been used with the multi-bit DACs.
The first of these employed the use of the brickwall filter. These filters had a very
sharp cutoff characteristic and held the signal gain close to unity almost to cutoff.
This was necessitated because the data was at a frequency such that aliasing and
noise artifacts existed immediately above the audible band. The inherent problem
with such a filter design was that they had tremendous phase nonlinearities at
high frequencies, and high-frequency group delaychange in phase shift with
respect to frequency. The second method of output reconstruction deals with an
oversampling digital filter prior to the DAC and a gentle analog filter. By gentle, it
is meant that a cutoff slope of 12 dB/octave and a -3 dB point of 30-40 kHz can be
used. Its design then is noncritical and low-orderwhich guarantees excellent
phase linearity. In fact, for most practical reconstruction filters, phase distortion
can be held at 0.5 over the entire audio band. The discussion of this is pertinent
to both multi- and low-bit DACs, so the topic will be covered after the next section.

L OW -BIT C ONVERTERS To combat the problems of multi-bit converters, two


competing technologies were developed, the first by Matsushita and the second by
Philips. Rather than converting whole data words in parallel at the sampling
frequency, both methods involve converting far shorter word lengths at far higher
rates. This serial data conversion is an inherently digital process and has been
made possible in part by the powerful digital signal processors available today.
12

Matsushitas method is based on pulse-width modulation (PWM). In this


design, the width of the signal pulse represents the unique data word, thus it is
critical that the PWM steps have exact width and minimum jitter to maximize
accuracy and linearity of the output. The commercial name for the process used
is MASH (Multi-stAge noise SHaping). A MASH converter is made of a 4-times
oversampling digital filter, followed by first- and second-order noise shapers in
parallel. The output from the noise shapers is then fed into a PWM converter,
whose output is then low-pass filtered. A block diagram of the MASH system is
shown in Fig. 2.

16-B IT I NTER - I NTER - 18-B IT


4X f s ATTENUATOR POLATING POLATING 4X f s
FIR FILTER FIR FILTER

L EFT
F IRST -O RDER +
+
POSITIVE L OW-PASS
NOISE LEFT
32 X fS L EFT F ILTER
SHAPER
N EGATIVE

SECOND - RIGHT
+
ORDER d POSITIVE L OW-PASS
R IGHT
NOISE dt RIGHT F ILTER
SHAPER
N EGATIVE

PWM D/A CONVERTER PERIPHERAL A UDIO CIRCUIT

Figure 2. Block diagram of a PWM/MASH digital-to-analog converter.

A digital finite impulse response (FIR) filter produces 18-bit data from a 16-bit
input sample after 4-times oversampling. The noise shapers then convert this 18-
bit data into an 11-step quantized format for the PWM after 8-times oversampling.
The PWM system is operated at 768 times the original sampling frequency (33.868
MHz). If it were to actually do a 1-bit conversion of 16-bit signals, 65,536 pulses
would be needed to represent each amplitude. However, this would require the
converter to operate at speeds in excess of 2.98 GHzfaster than the currently
available bipolar transistor technology. This restraint imposes the requirement
that the 18-bit data be reduced to the 11-step output. In practice the MASH
converter can only be considered a 3.5-bit converter.

The second low-bit conversion technique by Philips is known as pulse-density


modulation (PDM) or Bitstream conversion. In this technique, the density ratio of
the sign of the pulses is related to the original 16-bit data word. The PDM
converter is a true 1-bit technology. This signal representation may not seem
immediately obvious. A simple model helps illustrate what is happening4 . If a
13

light is on, then the room is brightly lit; if the light switch is off, the room is dark.
But if the switch is cycled rapidly on and off, an intermediate intensity can be
created. The sample data from the decoder chip is first passed to a low-pass non-
recursive 4-times oversampling FIR interpolation filter. This type of filter yields
higher quality because it is phase-linear. First-order noise shaping is performed
by the accumulator of the multiplier in the filter. The second filtering stage
consists of a 32-times oversampling linear interpolator and a 2-times
oversampling sample and hold circuit. At this stage, a 352 kHz digital dither
signal at -20 dB is added to the sample signal. This reduces nonlinearities
induced by quantization noise. At this point, the total oversampling is 256-times
and the data word has increased to 17-bits. The data is then fed at a frequency of
11.2896 MHz into the second order noise shaper. The noise shaper reduces the 17-
bit data to a 1-bit stream by using modulation. In this process quantization
noise is redistributed away from the audio frequency by as much as 2 orders of
magnitude. The bitstream is then converted to an analog form by a switched
capacitor network. A block diagram of the PDM converter is shown in Fig. 3.

16-BIT 4X O VERSAM PLIN G 1-BI T D/A T HI RD-O RDER L


44.1 K HZ (FIR F ILTER) (SW IT CHED CAPA CIT OR ANALO G AN AL OG OUTP U T
N ET WOR K ) L OW-PAS S FIL TER R

3 2X OVER SAMPL ING


(L INEAR INT ERPO LAT OR ) SEC OND -ORDE R
N OISE SHAPER
2X O VERSAM PLIN G
(SAMPL E AND HO LD)

Figure 3. Block diagram of a PDM digital-to-analog converter.

Because there are only two voltage references in the PDM converter, there is no
level matching requirement for improved accuracy. Therefore the linearity errors
associated with it are eliminated.

Comparisons of THD and linearity error for various 16-, 18-, 20-, and 1-bit
converters yield interesting results. PWM and PDM converters show < 1 dB
linearity for input signals from -100 to -80 dB and are virtually linear thereafter.
Some of the most expensive players on the market with 18- and 20-bit converters
using 4-, 8-, 16-, and even 32-times oversampling yield up to 4 dB linearity error
for signals as high as -75 dB. In the THD tests performed with a -60 dB 1 kHz sine
wave test signal, the expensive multi-bit players showed harmonics up to the 13th
14

at levels greater that -110 dB5. Only the PDM converter was able to hold all non-
fundamental harmonics under -110 dB.

Digital Filtering, Oversampling, and Noise Shaping


Oversampling is not mandated by any theorem discussed previously, but its use
yields tremendous performance gains regardless of the type of converter used.
Oversampling quite simply means using a sampling frequency greater than that
dictated by the Nyquist theorem. By exceeding the Nyquist frequency, many of the
precision demands made by the theorem can be relaxed (like the brickwall filter).
In addition to the benefits seen at the output filter, the signal-to-noise ratio is
boosted greatly and quantization noise is reduced in the audio band. The latter is
decreased by an incredible amount when oversampling is used in conjunction
with noise-shapers, which will be discussed shortly.

The oversampling process is well suited to a digital signal processor (DSP) ,


which essential takes in audio samples, performs an operation on them, and then
outputs audio samples. Because the samples are modified, the DSP is in effect a
digital filter. The DSP is beneficial because the operations it performs are precise
and repeatable, not otherwise possible with analog techniques, and result in lower
noise and distortion than with analog techniques. The oversampling process can
be viewed simply as interleaving zeros between each sample with additional
samples. In practice, these new samples are produced by using a shift register
(which acts as a delay line), coefficient multipliers, and an adder. The shift
register has taps after each delay element. The output of each tap is taken and
then multiplied by a coefficient stored in ROM associated with the impulse
response of the low-pass filter. These delayed multiples are then summed to
generate a new sample. An example of this can be seen in Fig. 4.
15

T1 T2 T3 T4
one- one- one- one-
Input
sample sample sample sample
(16-bit Samples
delay delay delay delay
@ 44.1 kHz)

Filter
Coefficients
C1 X C2 X C3 X C4 X

+ Output
@ n44.1kHz

Figure 4. Use of a transversal filter to achieve oversampling .

The total result of this process is that new interpolated samples are created at
each interleaved zero-value. This is shown graphically in Fig. 5.
SAMPLE VALUE

SAMPLE VALUE

SAMPLE VALUE

TIME TIME TIME


(a) (b) (c)
Figure 5. Effect of zeros interleaving and oversampling on a
signal. Original signal and samples (a) with: interleaved zeros
(b) and interpolated new samples (c).

As a result of this, the sampling frequency has increased by whatever amount of


oversampling occurred, and the data word length has grown. Because the
sampling frequency has risen, the noise in the audio band has been shifted out by
a greater amount than it was before. Noise shaping is then implemented to
reduce the data word size and further exaggerate the amount noise is moved out
of the audio band.

As stated previously, the primary job of the noise shaper is to alter the frequency
spectrum of the error signals so as to move most of the quantization error out of
the audible frequency range. Noise shaping reduces quantization noise by using a
negative feedback technique. In effect, the shaper attempts to reduce quantization
error by using its known qualities to actually subtract from the signal. The power
behind a low-bit conversion technique relies on the power of its noise-shaping
16

algorithm. In general, the more complex the noise-shaper, the lower the audio
band noise. Thus the performance of the noise shaper is determined by the order
of the shaper and its operating frequency. The latter parameter is a function of
how much oversampling is performed prior to shaping. The first relationship we
can extract from these parameters is the higher the order of the shaper, the
higher the slope of the noise redistribution and hence the lower the audible noise.
The drawback is that sideband noise is increased so much that the analog filters
could be overburdened. The second relationship is that the higher the operating
frequency, the higher in the frequency domain the noise is shifted. These two
relationships are defined by the noise-density distribution equation which is
shown in Eq. 4, where fs is the original sampling frequency and n is the shaper
order..
n
f
noise ( f ) = 2 sin (4)
fs

The relationship is also illustrated in Fig. 6a. The only limitation in operation
speed is the available speed of digital logic. Therefore, the conscientious designer
aims for the proper balance between shaping order and oversampling.

PWM / MASH: THIRD-ORDER NOISE SHAPER


8 AT 32X fs (1.4112 MHz) 0.012

0.010
6
RELATIVE NOISE DENSITY

RELATIVE NOISE DENSITY

0.008
PDM BITSREAM:
SECOND-ORDER NOISE SHAPER
AT 256X f s (11.2896 MHz)
4 0.006

PWM / MASH: THIRD-ORDER


NOISE SHAPER AT 32X
0.004
2

WITHOUT NOISE SHAPER 0.002 PDM BITSREAM:


SECOND-ORDER
NOISE SHAPER AT 256X

0 0

0 2 4 6 8 10 12 0 10 20 30 40 50
FREQUENCYMHz FREQUENCYkHz
(a) (b)
Figure 6. Various noise-density distributions as a function of frequency
(a). The PDM and PWM distributions in the audio band (b).
17

As a footnote, the operating frequency has the greatest effect of the two
parameters on noise density distribution. This is clearly visible in a much more
detailed look at the noise distributions in Fig. 6b. Clearly, the PDM has
significantly lower audio band noise and necessitates only a simple analog
reconstruction filter. A block diagram of the third-order noise shaper used in the
MASH converter is shown in Fig. 7.

FIRST-ORDER NOISE SHAPER

+ + Y(Z)
X(Z) +
Q1
INTEGRATOR

T
DELAY
d
+

+
dt

SECOND-ORDER NOISE SHAPER

+
+ +
+ Q2
INTEGRATOR INTEGRATOR

T
DELAY

Figure 7. Third-order noise shaper used in the PWM converter.

In the shaper given in Fig. 6, the input signal is fed into quantizer Q1 after the
residual error is subtracted from the delay block in the first order shaper. The
residual signal is also fed into the second order noise shaper, where the output of
the second quantizer, Q2, is differentiated and then summed with the output of
the first noise shaper to create the final output signal6.

The compact disc has only existed for about 13 years, and more than likely has
as many years of useful life left. There are many advances that are still possible in
the format and many of them are just in their infancy. However, many
challengers have already entered the playing field; some by the original creators
of the compact disc. Sony has created both the DAT standard as well as the Mini-
Disc, and Philips has created the DCC (digital compact cassette). Regardless of
18

the compact discs lifetime, it is certain that digital audio will remain, and analog
will be reserved to the role of input at the microphone in the studio and output at
the speaker in the listening environment.

This is by no means a complete or exhaustive analysis of the basic fundamentals


of the compact disc player. Many issues such as error-correction, data encoding
and decoding, and pickup design were neglected. However, the concepts covered
here should provide the reader with a strong background, and incite some
interest in learning more. For the reader who is interested in learning more, the
The Art of Digital Audio by Watkinson is an extensive collection of knowledge on
digital audio. It is at times very technical in nature, but the material introduced
builds upon itself nicely. Pohlmanns book, The Compact Disc Handbook, focuses
solely on the compact disc player and the compact disc itself along with all its
diverse formatsof which audio is only one. His book is very thorough in its
coverage and should leave no questions from the reader unanswered.
Pohlmanns book has a fair amount of overlap with Watkinsons and would make
a better starting point for those short on time.

ENDNOTES

1John Eargle. (Bitter Jitter and Sweeter Dither, Audio. Vol. 76, No. 1). p. 24

2Ken C. Pohlmann. (The Compact Disc HandbookThe Computer music and digital audio series.

2nd Ed. Madison, WI: A-R Editions, Inc. 1992). p. 34-35.

3Pohlmann, p. 46.

4Pohlmann, p. 148.

5Prasanna Shah. (Music of the Bitstream, Audio. Vol. 75, No. 1). p. 64.

6Shah, p. 61.
19

REFERENCES

Eargle, John. Bitter Jitter and Sweeter Dither, Audio. Vol. 76, No. 1 (Jan. 1992).
24+

Oppenheim, Alan V. Signals and Systems. Englewood Cliffs, NJ: Prentice-Hall,


Inc., 1983.

Pohlmann, Ken C. The Compact Disc HandbookThe Computer music and


digital audio series. 2nd Ed. Madison, WI: A-R Editions, Inc. 1992.

Shah, Prasanna. Music of the Bitstream, Audio. Vol. 75, No. 1 (Jan. 1991). 56 -
60+.

Strawn, John. Digital Audio EngineeringThe Computer music and digital


audio series. Los Altos, CA: William Kaufmann, Inc., 1985.

Strawn, John. Digital Audio Signal ProcessingThe Computer music and digital
audio series. Los Altos, CA: William Kaufmann, Inc., 1985.

Watkinson, John R. The Art of Digital Audio. 2nd Ed.. Boston, MA: Focal Press,
1994.

Watkinson, John R. Coding for Digital Recording. Boston, MA: Focal Press, 1990.

Whyte, Bert. Every Little Bit Helps, Audio. Vol. 76, No. 8 (Aug. 1992). 19-21.

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